|  | /* | 
|  | *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | * | 
|  | *  FEC and NACK added bitrate is handled outside class | 
|  | */ | 
|  |  | 
|  | #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 
|  | #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 
|  |  | 
|  | #include <stdint.h> | 
|  |  | 
|  | #include <deque> | 
|  | #include <memory> | 
|  | #include <utility> | 
|  | #include <vector> | 
|  |  | 
|  | #include "absl/types/optional.h" | 
|  | #include "api/field_trials_view.h" | 
|  | #include "api/network_state_predictor.h" | 
|  | #include "api/transport/network_types.h" | 
|  | #include "api/units/data_rate.h" | 
|  | #include "api/units/time_delta.h" | 
|  | #include "api/units/timestamp.h" | 
|  | #include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h" | 
|  | #include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h" | 
|  | #include "rtc_base/experiments/field_trial_parser.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class RtcEventLog; | 
|  |  | 
|  | class LinkCapacityTracker { | 
|  | public: | 
|  | LinkCapacityTracker(); | 
|  | ~LinkCapacityTracker(); | 
|  | // Call when a new delay-based estimate is available. | 
|  | void UpdateDelayBasedEstimate(Timestamp at_time, | 
|  | DataRate delay_based_bitrate); | 
|  | void OnStartingRate(DataRate start_rate); | 
|  | void OnRateUpdate(absl::optional<DataRate> acknowledged, | 
|  | DataRate target, | 
|  | Timestamp at_time); | 
|  | void OnRttBackoff(DataRate backoff_rate, Timestamp at_time); | 
|  | DataRate estimate() const; | 
|  |  | 
|  | private: | 
|  | FieldTrialParameter<TimeDelta> tracking_rate; | 
|  | double capacity_estimate_bps_ = 0; | 
|  | Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity(); | 
|  | DataRate last_delay_based_estimate_ = DataRate::PlusInfinity(); | 
|  | }; | 
|  |  | 
|  | class RttBasedBackoff { | 
|  | public: | 
|  | explicit RttBasedBackoff(const FieldTrialsView* key_value_config); | 
|  | ~RttBasedBackoff(); | 
|  | void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt); | 
|  | bool IsRttAboveLimit() const; | 
|  |  | 
|  | FieldTrialFlag disabled_; | 
|  | FieldTrialParameter<TimeDelta> configured_limit_; | 
|  | FieldTrialParameter<double> drop_fraction_; | 
|  | FieldTrialParameter<TimeDelta> drop_interval_; | 
|  | FieldTrialParameter<DataRate> bandwidth_floor_; | 
|  |  | 
|  | public: | 
|  | TimeDelta rtt_limit_; | 
|  | Timestamp last_propagation_rtt_update_; | 
|  | TimeDelta last_propagation_rtt_; | 
|  | Timestamp last_packet_sent_; | 
|  |  | 
|  | private: | 
|  | TimeDelta CorrectedRtt() const; | 
|  | }; | 
|  |  | 
|  | class SendSideBandwidthEstimation { | 
|  | public: | 
|  | SendSideBandwidthEstimation() = delete; | 
|  | SendSideBandwidthEstimation(const FieldTrialsView* key_value_config, | 
|  | RtcEventLog* event_log); | 
|  | ~SendSideBandwidthEstimation(); | 
|  |  | 
|  | void OnRouteChange(); | 
|  |  | 
|  | DataRate target_rate() const; | 
|  | LossBasedState loss_based_state() const; | 
|  | // Return whether the current rtt is higher than the rtt limited configured in | 
|  | // RttBasedBackoff. | 
|  | bool IsRttAboveLimit() const; | 
|  | uint8_t fraction_loss() const { return last_fraction_loss_; } | 
|  | TimeDelta round_trip_time() const { return last_round_trip_time_; } | 
|  |  | 
|  | DataRate GetEstimatedLinkCapacity() const; | 
|  | // Call periodically to update estimate. | 
|  | void UpdateEstimate(Timestamp at_time); | 
|  | void OnSentPacket(const SentPacket& sent_packet); | 
|  | void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt); | 
|  |  | 
|  | // Call when we receive a RTCP message with TMMBR or REMB. | 
|  | void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth); | 
|  |  | 
|  | // Call when a new delay-based estimate is available. | 
|  | void UpdateDelayBasedEstimate(Timestamp at_time, DataRate bitrate); | 
|  |  | 
|  | // Call when we receive a RTCP message with a ReceiveBlock. | 
|  | void UpdatePacketsLost(int64_t packets_lost, | 
|  | int64_t number_of_packets, | 
|  | Timestamp at_time); | 
|  |  | 
|  | // Call when we receive a RTCP message with a ReceiveBlock. | 
|  | void UpdateRtt(TimeDelta rtt, Timestamp at_time); | 
|  |  | 
|  | void SetBitrates(absl::optional<DataRate> send_bitrate, | 
|  | DataRate min_bitrate, | 
|  | DataRate max_bitrate, | 
|  | Timestamp at_time); | 
|  | void SetSendBitrate(DataRate bitrate, Timestamp at_time); | 
|  | void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate); | 
|  | int GetMinBitrate() const; | 
|  | void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate, | 
|  | Timestamp at_time); | 
|  | void UpdateLossBasedEstimator(const TransportPacketsFeedback& report, | 
|  | BandwidthUsage delay_detector_state, | 
|  | absl::optional<DataRate> probe_bitrate, | 
|  | bool in_alr); | 
|  |  | 
|  | private: | 
|  | friend class GoogCcStatePrinter; | 
|  |  | 
|  | enum UmaState { kNoUpdate, kFirstDone, kDone }; | 
|  |  | 
|  | bool IsInStartPhase(Timestamp at_time) const; | 
|  |  | 
|  | void UpdateUmaStatsPacketsLost(Timestamp at_time, int packets_lost); | 
|  |  | 
|  | // Updates history of min bitrates. | 
|  | // After this method returns min_bitrate_history_.front().second contains the | 
|  | // min bitrate used during last kBweIncreaseIntervalMs. | 
|  | void UpdateMinHistory(Timestamp at_time); | 
|  |  | 
|  | // Gets the upper limit for the target bitrate. This is the minimum of the | 
|  | // delay based limit, the receiver limit and the loss based controller limit. | 
|  | DataRate GetUpperLimit() const; | 
|  | // Prints a warning if `bitrate` if sufficiently long time has past since last | 
|  | // warning. | 
|  | void MaybeLogLowBitrateWarning(DataRate bitrate, Timestamp at_time); | 
|  | // Stores an update to the event log if the loss rate has changed, the target | 
|  | // has changed, or sufficient time has passed since last stored event. | 
|  | void MaybeLogLossBasedEvent(Timestamp at_time); | 
|  |  | 
|  | // Cap `bitrate` to [min_bitrate_configured_, max_bitrate_configured_] and | 
|  | // set `current_bitrate_` to the capped value and updates the event log. | 
|  | void UpdateTargetBitrate(DataRate bitrate, Timestamp at_time); | 
|  | // Applies lower and upper bounds to the current target rate. | 
|  | // TODO(srte): This seems to be called even when limits haven't changed, that | 
|  | // should be cleaned up. | 
|  | void ApplyTargetLimits(Timestamp at_time); | 
|  |  | 
|  | bool LossBasedBandwidthEstimatorV1Enabled() const; | 
|  | bool LossBasedBandwidthEstimatorV2Enabled() const; | 
|  |  | 
|  | bool LossBasedBandwidthEstimatorV1ReadyForUse() const; | 
|  | bool LossBasedBandwidthEstimatorV2ReadyForUse() const; | 
|  |  | 
|  | const FieldTrialsView* key_value_config_; | 
|  | RttBasedBackoff rtt_backoff_; | 
|  | LinkCapacityTracker link_capacity_; | 
|  |  | 
|  | std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_; | 
|  |  | 
|  | // incoming filters | 
|  | int lost_packets_since_last_loss_update_; | 
|  | int expected_packets_since_last_loss_update_; | 
|  |  | 
|  | absl::optional<DataRate> acknowledged_rate_; | 
|  | DataRate current_target_; | 
|  | DataRate last_logged_target_; | 
|  | DataRate min_bitrate_configured_; | 
|  | DataRate max_bitrate_configured_; | 
|  | Timestamp last_low_bitrate_log_; | 
|  |  | 
|  | bool has_decreased_since_last_fraction_loss_; | 
|  | Timestamp last_loss_feedback_; | 
|  | Timestamp last_loss_packet_report_; | 
|  | uint8_t last_fraction_loss_; | 
|  | uint8_t last_logged_fraction_loss_; | 
|  | TimeDelta last_round_trip_time_; | 
|  |  | 
|  | // The max bitrate as set by the receiver in the call. This is typically | 
|  | // signalled using the REMB RTCP message and is used when we don't have any | 
|  | // send side delay based estimate. | 
|  | DataRate receiver_limit_; | 
|  | DataRate delay_based_limit_; | 
|  | Timestamp time_last_decrease_; | 
|  | Timestamp first_report_time_; | 
|  | int initially_lost_packets_; | 
|  | DataRate bitrate_at_2_seconds_; | 
|  | UmaState uma_update_state_; | 
|  | UmaState uma_rtt_state_; | 
|  | std::vector<bool> rampup_uma_stats_updated_; | 
|  | RtcEventLog* const event_log_; | 
|  | Timestamp last_rtc_event_log_; | 
|  | float low_loss_threshold_; | 
|  | float high_loss_threshold_; | 
|  | DataRate bitrate_threshold_; | 
|  | LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_; | 
|  | std::unique_ptr<LossBasedBweV2> loss_based_bandwidth_estimator_v2_; | 
|  | LossBasedState loss_based_state_; | 
|  | FieldTrialFlag disable_receiver_limit_caps_only_; | 
|  | }; | 
|  | }  // namespace webrtc | 
|  | #endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |